Re: [asterisk-users] Blind Transfer not working - 1.4.38

2011-01-14 Thread Ishfaq Malik 
This is a heads up to everyone

Apparently this is a known but in the latest version on asterisk 1.4,
1.6 and 1.8

http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1-6-2-15-and-1-8-0-1

https://issues.asterisk.org/view.php?id=18185

On Thu, 2011-01-06 at 13:10 +, Ishfaq Malik wrote:
 On Wed, 2011-01-05 at 15:47 +, Ishfaq Malik wrote:
  Hi
  
  We've been running asterisk 1.4.17 (deb package) in a production
  environment for some while now and are finally taken the plunge to
  update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
  Architecture 
  
  I have upgraded the asterisk version in one of our test environments and
  blind transferring seems to have suddenly stopped working. It was
  working fine under 1.4.17
  
  So, call comes in to extension 501 who does a blind transfer to
  extension 504 at which point the call gets completely cut off.
  
  I ran a SIP trace of this happening and it appears to be attempting to
  do the transfer:
  
  -
  --- (12 headers 0 lines) ---
  Call 7c5d5a603b2803fd7e451de826e4@x.x.x.x got a SIP call transfer from 
  caller: (REFER)!
  SIP transfer to extension 504@pack-local by pack...@domain.co.uk
  
  --- Transmitting (NAT) to x.x.x.x:52753 ---
  SIP/2.0 202 Accepted
  Via: SIP/2.0/UDP 
  192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rport=52753
  From: sip:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
  To: incoming mobile number sip:incoming mobile 
  number@x.x.x.x;tag=as4d0dbc04
  Call-ID: 7c5d5a603b2803fd7e451de826e4@x.x.x.x
  CSeq: 2 REFER
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Contact: sip:incoming mobile number@x.x.x.x
  Content-Length: 0
  
  
  
  set_destination: Parsing sip:PACK501@192.168.1.105:3072;line=guuuyf05 for 
  address/port to send to
  set_destination: set destination to 192.168.1.105, port 3072
  Reliably Transmitting (NAT) to x.x.x.x:52753:
  NOTIFY sip:PACK501@192.168.1.105:3072;line=guuuyf05 SIP/2.0
  Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
  From: incoming mobile number sip:incoming mobile 
  number@x.x.x.x;tag=as4d0dbc04
  To: sip:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
  Contact: sip:incoming mobile number@x.x.x.x
  Call-ID: 7c5d5a603b2803fd7e451de826e4@87.237.58.231
  CSeq: 103 NOTIFY
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Remote-Party-ID: incoming mobile number sip:incoming mobile 
  number@x.x.x.x;privacy=off;screen=no
  Event: refer;id=2
  Subscription-state: active
  Content-Type: message/sipfrag;version=2.0
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Content-Length: 21
  
  SIP/2.0 183 Ringing
  
  
  ___
  But as stated above, extension 504 doesn't ring and the call dies.
  
  
  Now 504 is a valid extensions in the context pack-local
  select * from extensions where exten='_5XX';
  +---++---+--+---+---+
  | id| context| exten | priority | app   | appdata   
  |
  +---++---+--+---+---+
  | 65127 | pack-local | _5XX  |1 | Macro | 
  stdexten|${EXTEN}|pack-local|PACK | 
  +---++---+--+---+---+
  
  
  Also, attended transfers work without a problem.
  
  Both SIP phones used were Snom phones.
  
  Has anyone encountered an issue like this before?
  
  
 
 I spotted something new here, when I try to do the blind transfer I get
 the following output on the console
 
 == Spawn extension (pack-local, 504, 0) exited non-zero on
 
 So why would it be looking at priority 0 rather than priority 1?
 
 -- 
 Ishfaq Malik
 Software Developer
 PackNet Ltd
 
 Office:   0161 660 3062
 
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Blind Transfer not working - 1.4.38

2011-01-14 Thread Mike 
Hi,

1.6.2.16rc1 does not have this problem (that`s why I am running a release
candidate right now).  Can`t say about 1.4 versions, but it`s safe to say
whatever they fixed will be out in the next version.

Mike

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, January 14, 2011 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind Transfer not working - 1.4.38

This is a heads up to everyone

Apparently this is a known but in the latest version on asterisk 1.4,
1.6 and 1.8

http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1
-6-2-15-and-1-8-0-1

https://issues.asterisk.org/view.php?id=18185

On Thu, 2011-01-06 at 13:10 +, Ishfaq Malik wrote:
 On Wed, 2011-01-05 at 15:47 +, Ishfaq Malik wrote:
  Hi
  
  We've been running asterisk 1.4.17 (deb package) in a production 
  environment for some while now and are finally taken the plunge to 
  update to 1.4.38 (Ubuntu servers). All of this is using the RealTime 
  Architecture
  
  I have upgraded the asterisk version in one of our test environments 
  and blind transferring seems to have suddenly stopped working. It 
  was working fine under 1.4.17
  
  So, call comes in to extension 501 who does a blind transfer to 
  extension 504 at which point the call gets completely cut off.
  
  I ran a SIP trace of this happening and it appears to be attempting 
  to do the transfer:
  
  -
  --- (12 headers 0 lines) ---
  Call 7c5d5a603b2803fd7e451de826e4@x.x.x.x got a SIP call transfer
from caller: (REFER)!
  SIP transfer to extension 504@pack-local by pack...@domain.co.uk
  
  --- Transmitting (NAT) to x.x.x.x:52753 ---
  SIP/2.0 202 Accepted
  Via: SIP/2.0/UDP 
  192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rpor
  t=52753
  From: sip:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
  To: incoming mobile number sip:incoming mobile 
  number@x.x.x.x;tag=as4d0dbc04
  Call-ID: 7c5d5a603b2803fd7e451de826e4@x.x.x.x
  CSeq: 2 REFER
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
  INFO
  Supported: replaces
  Contact: sip:incoming mobile number@x.x.x.x
  Content-Length: 0
  
  
  
  set_destination: Parsing 
  sip:PACK501@192.168.1.105:3072;line=guuuyf05 for address/port to 
  send to
  set_destination: set destination to 192.168.1.105, port 3072 
  Reliably Transmitting (NAT) to x.x.x.x:52753:
  NOTIFY sip:PACK501@192.168.1.105:3072;line=guuuyf05 SIP/2.0
  Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
  From: incoming mobile number sip:incoming mobile 
  number@x.x.x.x;tag=as4d0dbc04
  To: sip:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
  Contact: sip:incoming mobile number@x.x.x.x
  Call-ID: 7c5d5a603b2803fd7e451de826e4@87.237.58.231
  CSeq: 103 NOTIFY
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Remote-Party-ID: incoming mobile number sip:incoming mobile 
  number@x.x.x.x;privacy=off;screen=no
  Event: refer;id=2
  Subscription-state: active
  Content-Type: message/sipfrag;version=2.0
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
  INFO
  Supported: replaces
  Content-Length: 21
  
  SIP/2.0 183 Ringing
  
  
  
  ___
  But as stated above, extension 504 doesn't ring and the call dies.
  
  
  Now 504 is a valid extensions in the context pack-local select * 
  from extensions where exten='_5XX';
 
+---++---+--+---+---
+
  | id| context| exten | priority | app   | appdata
|
 
+---++---+--+---+---
+
  | 65127 | pack-local | _5XX  |1 | Macro |
stdexten|${EXTEN}|pack-local|PACK | 
 
+---++---+--+---+---
+
  
  
  Also, attended transfers work without a problem.
  
  Both SIP phones used were Snom phones.
  
  Has anyone encountered an issue like this before?
  
  
 
 I spotted something new here, when I try to do the blind transfer I 
 get the following output on the console
 
 == Spawn extension (pack-local, 504, 0) exited non-zero on
 
 So why would it be looking at priority 0 rather than priority 1?
 
 --
 Ishfaq Malik
 Software Developer
 PackNet Ltd
 
 Office:   0161 660 3062
 
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Ishfaq Malik
Software Developer

Re: [asterisk-users] Blind Transfer not working - 1.4.38

2011-01-06 Thread Ishfaq Malik 
On Wed, 2011-01-05 at 15:47 +, Ishfaq Malik wrote:
 Hi
 
 We've been running asterisk 1.4.17 (deb package) in a production
 environment for some while now and are finally taken the plunge to
 update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
 Architecture 
 
 I have upgraded the asterisk version in one of our test environments and
 blind transferring seems to have suddenly stopped working. It was
 working fine under 1.4.17
 
 So, call comes in to extension 501 who does a blind transfer to
 extension 504 at which point the call gets completely cut off.
 
 I ran a SIP trace of this happening and it appears to be attempting to
 do the transfer:
 
 -
 --- (12 headers 0 lines) ---
 Call 7c5d5a603b2803fd7e451de82...@x.x.x.x got a SIP call transfer from 
 caller: (REFER)!
 SIP transfer to extension 5...@pack-local by pack...@domain.co.uk
 
 --- Transmitting (NAT) to x.x.x.x:52753 ---
 SIP/2.0 202 Accepted
 Via: SIP/2.0/UDP 
 192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rport=52753
 From: sip:pack...@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
 To: incoming mobile number sip:incoming mobile 
 number@x.x.x.x;tag=as4d0dbc04
 Call-ID: 7c5d5a603b2803fd7e451de82...@x.x.x.x
 CSeq: 2 REFER
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces
 Contact: sip:incoming mobile number@x.x.x.x
 Content-Length: 0
 
 
 
 set_destination: Parsing sip:pack...@192.168.1.105:3072;line=guuuyf05 for 
 address/port to send to
 set_destination: set destination to 192.168.1.105, port 3072
 Reliably Transmitting (NAT) to x.x.x.x:52753:
 NOTIFY sip:pack...@192.168.1.105:3072;line=guuuyf05 SIP/2.0
 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
 From: incoming mobile number sip:incoming mobile 
 number@x.x.x.x;tag=as4d0dbc04
 To: sip:pack...@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
 Contact: sip:incoming mobile number@x.x.x.x
 Call-ID: 7c5d5a603b2803fd7e451de82...@87.237.58.231
 CSeq: 103 NOTIFY
 User-Agent: Asterisk PBX
 Max-Forwards: 70
 Remote-Party-ID: incoming mobile number sip:incoming mobile 
 number@x.x.x.x;privacy=off;screen=no
 Event: refer;id=2
 Subscription-state: active
 Content-Type: message/sipfrag;version=2.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces
 Content-Length: 21
 
 SIP/2.0 183 Ringing
 
 
 ___
 But as stated above, extension 504 doesn't ring and the call dies.
 
 
 Now 504 is a valid extensions in the context pack-local
 select * from extensions where exten='_5XX';
 +---++---+--+---+---+
 | id| context| exten | priority | app   | appdata 
   |
 +---++---+--+---+---+
 | 65127 | pack-local | _5XX  |1 | Macro | 
 stdexten|${EXTEN}|pack-local|PACK | 
 +---++---+--+---+---+
 
 
 Also, attended transfers work without a problem.
 
 Both SIP phones used were Snom phones.
 
 Has anyone encountered an issue like this before?
 
 

I spotted something new here, when I try to do the blind transfer I get
the following output on the console

== Spawn extension (pack-local, 504, 0) exited non-zero on

So why would it be looking at priority 0 rather than priority 1?

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Blind Transfer not working - 1.4.38

2011-01-05 Thread Ishfaq Malik 
Hi

We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture 

I have upgraded the asterisk version in one of our test environments and
blind transferring seems to have suddenly stopped working. It was
working fine under 1.4.17

So, call comes in to extension 501 who does a blind transfer to
extension 504 at which point the call gets completely cut off.

I ran a SIP trace of this happening and it appears to be attempting to
do the transfer:

-
--- (12 headers 0 lines) ---
Call 7c5d5a603b2803fd7e451de82...@x.x.x.x got a SIP call transfer from 
caller: (REFER)!
SIP transfer to extension 5...@pack-local by pack...@domain.co.uk

--- Transmitting (NAT) to x.x.x.x:52753 ---
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 
192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rport=52753
From: sip:pack...@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
To: incoming mobile number sip:incoming mobile 
number@x.x.x.x;tag=as4d0dbc04
Call-ID: 7c5d5a603b2803fd7e451de82...@x.x.x.x
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:incoming mobile number@x.x.x.x
Content-Length: 0



set_destination: Parsing sip:pack...@192.168.1.105:3072;line=guuuyf05 for 
address/port to send to
set_destination: set destination to 192.168.1.105, port 3072
Reliably Transmitting (NAT) to x.x.x.x:52753:
NOTIFY sip:pack...@192.168.1.105:3072;line=guuuyf05 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
From: incoming mobile number sip:incoming mobile 
number@x.x.x.x;tag=as4d0dbc04
To: sip:pack...@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
Contact: sip:incoming mobile number@x.x.x.x
Call-ID: 7c5d5a603b2803fd7e451de82...@87.237.58.231
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: incoming mobile number sip:incoming mobile 
number@x.x.x.x;privacy=off;screen=no
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 21

SIP/2.0 183 Ringing


___
But as stated above, extension 504 doesn't ring and the call dies.


Now 504 is a valid extensions in the context pack-local
select * from extensions where exten='_5XX';
+---++---+--+---+---+
| id| context| exten | priority | app   | appdata   
|
+---++---+--+---+---+
| 65127 | pack-local | _5XX  |1 | Macro | 
stdexten|${EXTEN}|pack-local|PACK | 
+---++---+--+---+---+


Also, attended transfers work without a problem.

Both SIP phones used were Snom phones.

Has anyone encountered an issue like this before?


-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users